PRINCIPLES OF DESIGNING A N ALGORITHM FOR ACOUSTIC ROOM MULTICHANNEL SIMULATION

Miroslav Balik
Dept. of Telecommunications Faculty of Electrical Engineering and Communication Technologies Brno University of Technology Purkyitova 118,612 00 Brno, Czech Republic Phone: +420-5-41149190,Fax: +420-5-41149192 Email: balik@feec.vutbr.cz

ABSTRACT
The paper deals with the principles of realizing a multichannel simulation of acoustic room (known as an artificial reverberation) as a discrete system. The resulting simulation algorithm describing the discrete system is to be adapted to the realization in a DSP environment and should be capable of working in real time. A definition of the requirements made on perceptual approach of subsequent reverberation is part of the paper. In conclusion, two models for multichannel reverberation are proposed. The models are designed to have the best possible relation between simulation quality and computation requirements.

1.. INTRODUCTION
Digital processing of audio signal is currently undergoing a rapid development. This has been made possible by the extremely rapid development in the area of processor systems. Until recently, dedicated DSP systems were mostly used for the realization of algorithms for digital processing of audio signals. These dedicated DSP systems are comparatively costly and for any further specific algorithm another dedicated DSP system must be used. The present trend is to realize the audio signal processing algorithms on the CPU of a high-performance PC. The steeply rising CPU performance of current PCs makes it possible to realize complete mixing, effect and mastering software for audio signal processing. A typical example can be seen in the HDR systems. No further specialized DSP system is required for its fully satisfactory function, an internal sound card or a professional external D/A converter is sufficient. This paper. describes the principles of designing an algorithm for the simulation of the properties of an acoustic room - so-called reverberation algorithm or artificial reverberation. The theory of these algorithms has been described comparatively well but the results obtained do not always fully attest the theory. The fundamental problem consists in the very specification of the requirements made on this algorithm. The algorithm itself and the demands made on it can hardly be described

with mathematical accuracy (this is true of the perceptual description, which we are going to deal with). The main contribution of this paper should be a specification of the requirements made on the reverberation algorithm, which will be prepared for real-time processing and which will meet the requirement for multichannel output. The multichannel output of the reverberation algorithm is mostly required in mastering studios in the production of a DVD master. It finds application mainly in positioning the source of sound in a certain acoustic room (an ordinary room, streets in the town, corridors, etc.). Currently, the following numbers of output channels are considered: 2, 3,4, 5 and 7. These numbers correspond to present-day standards for multichannel playback (see [ 11[21).

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Fig. 1. Possible multichannel speaker placement (i.e. 7.1 playback) with denoted reverberator output signals (see chapter 5.)

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left front speaker center speaker right front speaker let? surround speaker right surround speaker left back surround speaker right back surround speaker

DESCRIPTION OF ACOUSTIC ROOM PROPERTIES

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Simulating an acoustic room means finding a system that describes the properties of sound beam propagation from the source of sound towards the listener in an open

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or closed room. The following physical parameters are of hdamental effect on the properties of this system - the size of the closed room, its geometrical lay-out, the material that the walls of and the objects in the closed room are made of, the position of the source of sound and the listener's position (in some models also the position of the head).

Fig. 2. DIR of MLSSA mesuring system in an anechoic room (DIR of whole electroacoustic chain included)

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Fig. 3. DIR of hallway at our department (microphone and speaker are oriented to each other)

Fig. 4. DIR of hallway at our department (microphone and speaker are oriented from each other) The reverberation algorithms are based on either the physical or the perceptual approach of the acoustic room properties (see [3]). The physical approach is given by some physical property of the acoustic room. The impulse characteristic is used most frequently. Measuring the impulse characteristic of an acoustic room has its pitfalls; ideally,

it describes the behaviour of sound rays as they travel from a multidirection point source to a multidirection point receiver. This is impossible in practice since the transmitter is a loudspeaker and the receiver I S a microphone and neither of them can be regarded as a multidirection point source or receiver. However, if certain conditions are satisfied and certain properties of the room are neglected while measuring the impulse response (IR or its discrete version DIR), the measuring can be regarded as a feasible physical description of an acoustic room. The MLSSA system, based on the MLS theory, is usually used for the measuring (see [4]). The measured DIR then describes quite exactly the digital system that can be used to simulate the properties of an acoustic room with a specific source and receiver layout. But the realization of the algorithm is a problem: computing the convolution or even the fast convolution is still so demanding that it cannot be realized on current PC systems in real-time. Typical response lengths of large acoustic rooms with decay times about 2,Ss exceed the value of 100 000 (N=110250) samples at F,=44.1 icHz. The perceptual approach starts from the sensory perception of a sound that is being played back foi the listener in the respective acoustic room. The perceptual approach aims at reducing the computation requirements of the simulation algorithm. The algorithm does not simulate the IR of an acoustic room directly, it only simulates the specific features of an acoustic room. These specific features are defined as human perceptions. Let the room of all the perceptions caused by sound ray propagation in an acoustic room be distributed into D dimensions that would correspond to all independent perceptions caused by sound ray propagation. If each of the perceptions can be described by a definite physical property of the acoustic room a digital filter can be designed with D parameters, which accurately simulate each of the D independent perceptions. In comparison with the physical approach, the perceptual approach has several advantages. Acoiistic room simulation using perceptual approach is based on a much more effective algorithm - on delay lines and filters of the type of IIR. This results in substantially lower computation requirements than in the case of the physical model, where type FIR filters are employed. The model enables altering all the D parameters in real time and, moreover, neither these parameters nor their changes need be correlated, as is the case of the physical model. The changes in model parameters are applied without noticeable delay, in contrast to the physical model, where a change in parameters shows only after as many samples as correspond to the length of the acoustic room IR. An disadvantage, however, can be seen in that using the perceptual model we cannot simulate the given acoustic room precisely, we can only approximate its properties.

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In the perceptual model structure two basic blocks can in most cases be distinguished, which are based on completely different algorithms. These are the block that simulates early reflections of the sound ray and the block simulating subsequent reverberation. The present paper deals in greater detail only with the block that simulates subsequent reverberation of sound beams in an acoustic room.
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e Fig. 5. Echogram of an general reverberation algorithm based on perceptual approach The Fig. 5 shows in principle the echogram of an acoustic room simulator. From the specifications of basic parameters the part of early reflections and the part of subsequent reverberation are given. The IR of early reflections is intentionally shown as sparse (IR in time acquires mostly zero values). The Fig. 6 gives the actual DIR of an algorithm based on perceptual approach. The DIR shown corresponds to the output of one of the reverberation algorithm channels, which is described in conclusion of this paper. At first sight, the DIR parameters are very similar to actual DIRs measured (see Fig.2,3,4,5). The parameters are described in the next chapter.
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channel reverberator are determines on the basis of D parameters, which may include: the ratio of the original and the processed signal the ratio of the output signal of the early reflections block and the subsequent reverberation block athe mutual delay between the output signals of individual blocks and the input signal the size of acoustic room described by the decay time the shape of acoustic room defined by another reflection distribution the color - it describes the amount of absorption of a certain part of the frequency spectrum of sound beams that originates on material with frequency-dependent absorptivenessduring beam propagation in a room the absorption is modelled by frequency filters placed in the direct path of effected signal the absorption is modelled by a frequency-dependent feedback, i.e. frequency filters placed in the system's feedback the density of reflections in the acoustic room, the number of non-zero values in the discrete IR possibly some others, for more exacting requirements. The algorithm is to provide these parameters for the user and, by setting them, the user should have such sound perceptions as if he were listening to the given sound recording in real time. The requirement is for all these parameters to be adjustable during signal processing in real time and their changes within a single-channel simulator should be uncorrelated. Some of these parameters are comparatively easy to simulate independently of the type of algorithm, others are given directly by the type of algorithm. The greatest problem may be expressing the parameters of reflection density and distribution. The density and distribution of reflections in the acoustic room are functions of time, for example it holds for the density that it grows with time. Another important requirement is the simulation . capability of working in real time. In postproduction work in HDR studios the requirements made on the speed of algorithm calculation are greater than those made on the capability to process in real time. In this case the degree of algorithm computation requirements is described by the loading of the digital system on which the algorithm is being processed. The degree is given as a percentage, with 100% corresponding to the maximum possible loading of the digital system. For a given digital system it holds that the algorithm which loads the system 100% is the most computation-demanding algorithm that the given digital system can process in real time. In practice this means that an algorithm taking up 25% of a given digital system is processed four times faster than in real time. A common requirement in HDR studios the simultaneous processing of a signal or signals by several algorithms. If the algorithms require much computation

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Fig. 6. DIR of common reverberator based on perceptual approach

3.

REVERBERATOR REQUIREMENTS

Determining the demands made on reverberator properties is a comparatively complicated task if we start from the perceptual approach. Unlike the physical approach, where the reverberator requirement is its DIR, in the case of the perceptual approach the requirements are given by the listener's subjective impressions, which need not be unambiguous. The requirements for a single-

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and a sufficientlypowerful digital system is not available, the postproduction time is disproportionately high. The resulting algorithm should be maximally optimized to load the digital system as little as possible but at the same time to meet the demands made on it. A practical benefit resulting from this requirement is a considerable reduction of the time necessary for postproduction.

3.1.

Requirements on a discrete system for subsequent reverberation

If we neglect the frequency-dependent absorptiveness of material in an acoustic room and eventually absorptiveness as such (if a material is sued that reflects the sound beam without absorbing part of its energy), we speak of an acoustic room with ideal sound beam reflectivity. This is in fact the opposite of so-called anechoic acoustic room. The discrete system for the simulation of subsequentreverberation can then be defined as a discrete system whose DIR is a signal with white noise properties and with the number of non-zero impulses in the DIR increasing with time. The discrete system is solely of the type of IIR with comparatively complicated feedbacks and, consequently, with a relatively complicated specificationof the stability criterion. If there is such a discrete system that fulfils the above assumption, it is relatively easy to adapt it for the simulation of real acoustic room properties. The measure of absorptiveness is then simulated by the measure of the negative feedbacks smaller than 1. The frequency dependence of absorptivenessis introduced by frequency filters in both the direct signal path and the discrete system feedbacks. For the simulation to be successful,the absorptiveness of commonly used materials must be known. For a given type of acoustic room it is necessary to weight the frequency responses of the filters applied, in dependenceon the material present in the acoustic room. The requirements on one output signal from a multichannel system for the simulation of subsequent reverberation are identical to those made on the output signal from a single-channel system described above. It is only necessary to define the ration of output signals. The situation when the output signals from a multichannels system are uncorrelated is regarded as ideal.
4.

.BASIC METHODS FOR SIMULATING MULTIPLE REFLECTIONS

The first algorithms for the simulation of acoustic room properties were published in the 1961 (see [5]). Since then the algorithms have gone a long way. In the course of their development the quality of simulation has been improved and, moreover, the requirements made on this simulation have been specified. In the following, a

survey of the structures used for subsequent reverheration simulation is given: *Parallel comb filters - used by Schroeder for the acoustic room sirnulation (see []. A sufficient inumber 6) of parallel-arranged comb filters is to simulate a sufficientnumber of eigen frequencies in a closed room. To achieve a sufficient density of non-zero samples in DIR, either a high number of parallel comb filters is chosen or a series of all-pass filters, which increase the frequency of non-zero samples in DIR, are placed after a lower number of parallel comb filters. This algorithm does not enable increasing the density of non-zero samples with time. The parameters of the algorithm are established empirically. *All-pass filter series with feedbacks - used in the Dattorroos reverberator (see [7]); a number of all-pass filters are connected in series and through some of them the feedback is introduced.The DIR density is sufficient and it does not increase with time. Due to the feedback introduced, periodic fluctuation in the signal level, which is similar to amplitude modulation by a low frequency, can be observed in the output algorithm. The design of the algorithm parameters is very complicated and difficult to determine. Nested all-pass filters (NAP) - used in the Gardners reverberation structures (see [ ] ; unlike the all-pass 8) filter, the NAP includes in addition to a delay block a fiuther all-pass filter. An advantage of this structure is that the DIR density increases with time, as in real conditions. The property of the all-pass filter is maintained - the module of frequency respoase is constant. If a great number of NAPS are combined, the design of the algorithm parameters is relatively complicated and difficult to determine. *Feedback delay networks (FDN) - based on an extensive theory and used in the Jots reverberation networks. The network design has been described in detail and in theory it should meet the requirements made on subsequent reverberation (see 191). The feedback matrix can be designed such that it satisfies the stability criterion and that the system with these matrix feedbacks has a clearly defined length of DIR, in contrast to the Dattorros and the Gardners structures. The design of input and output weighting coefficients, which are connected to the inputs and outputs of delay blocks with matrix feedback, is problematic. Circulant feedback delay networks (CFDN) - used in reverberators by Rocchesso and Smith (see [lo]:,. This is a special kind of FDN with cyclic changes in the coefficients in the feedback matrix. They have properties very similar to FDN, an advantage is the possibility of using the FFT in the calculation algcdthm. In CFDN the order of the feedback matrix can be higher than in FDN with the same computation requirements. The algorithm as such is comparatively complicatvd.

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*Digital waveguide networks (DWN) - in principle they behave as in FDN but they are relatively demanding on discrete system simulation.
5.

DESIGN OF COLORLESS SUBSEQUENT REVERBERATOR

Before starting to design the reverberator it is necessary to study in detail the structures given above and to become familiar with their features. DIR and frequency response are of special importance. As we know, the requirement on subsequent reverberation is a sufficient frequency of non-zero values, which increases with time, in the DIR. Another important requirement is the whiteness of DIR, which is necessary for so-called colorless reverberation. This requirement is much more difficult to meet. It is easy to derive theoretically that the module of frequency response of an all-pass filter is a constant function. In practice this is easy to verify by calculating DFT from a sufficient number of samples of DIR (the system is of the IIR type and regarded as a sufficient number of DIR samples is the number that correspondsto the definition of decay time). It could be expected that the module of the spectrum of a signal that has been filtered by an all-pass filter will in no way be affected in comparison with the module of the spectrum of the input signal. This of course only holds in the case of signal frequency analysis over the whole length interval of DIR. The frequency properties of the human ear can be compared to the frequency analyser of short-time spectra. If we simulate this behaviour by an algorithm for the calculation of the short-time spectrum of a signal and analyse a signal that has been filtered by an all-pass filter, we find that the individual short-time modules of the spectrum have been affected a lot. The all-pass filter as such cannot therefore be regarded as the algorithm for colorless reverberation. The point is to obtain such properties of the reverberator DIR that the short-time modules of the spectrum of the passing signal are in no way affected by the reverberator. It only remains to determine the time interval for the calculation of shorttime spectrum, which is defined by the perception inertia of the human ear. Procedures for satisfying the requirements mad on the reverberator are comparatively numerous but they are not computation-effective. The design criterion is thus a balance between satisfjmg the requirements and the computation demands.
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been optimized in respect of the computation performace but it does. not meet the requirement for uncorrelated output from the multichannel subsequent reverberator, with the output signals formed by a linear combination of a set of signals. The Model(2) is much more demanding, it meets all the requirements made on the reverberator and, moreover, the absorption properties of the materials in the acoustic room can be defined with greater accuracy.
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Fig. 7. Block diagram of Model( 1)

MULTICHANNEL REVERBERATOR, THEORETICAL DESIGN

In conclusion I describe two potential solutions of multichannel reverberator that have been proposed on the basis of the criterions given above. The Model(1) has

DL (Delay Line) - delay line with N outputs. Each value of delay is specified by significant sound beam reflection in 3-D simplified acoustic room model. N should be equal to values ranging from 20 to 50 for sufficient initial early reflections (ER) density. AB (Absorption filters) - each delay line output has its IIR absorption filter (typically first or second order systems); for increasing density of ER N all-pass filters with short decay time are used. LSM (Nx2)Winear Scaling Matrix) - 2 outputs from LSM are defined as 2 weighted sums of N inputs. The output represents the Part1 and Part2 of ER shown on Fig. 5, but not exactly as it is in Model(2). NAP (Nested Allpass Filters) - 1 to 4 NAPS are used for each channel (2 to 8) to highly increase the initial subsequent reverberation (SR) density PCF (Parallel Comb Filters) - most demanding part of the algorithm - P comb filters with frequency dependent feedbacks to simulate eigen frequencies of acoustic room are used. For single channel SR is necessary at least 8 to 12 CF. For S i to S6 uncorelated outputs should be used 48 to 72 CF. To reduce computational complexity about 16 and more CF are necessary for natural sounding reverberation, but the S1 to S6 output will be corelated LSM (PxL)(Linear Scaling Matrix) - this block produces 6 subsequent reverberation outputs S1 to S6

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(surround sounds). They are defined as weighted sums of P outputs from PCF.
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these signals go to format conversion module, where they are converted to desired audio output format (see [12, 11).

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CONCLUSION

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This paper presented principles of designing an algorithm for multichannel artficial reverberation. The DS-RVl Reverb DirectX and VST compatible plug-in based on these principles have been designed. Computation requirement of DS-RVl is only about 4% on PC with AMD 1333MHz when processing a stereo signal. The Model(1) with stereo output has been used (N=20, P=16). The 8-channel VST plug-in for NUEIDO DAW is currently under development. The demo version of DS-RV1 Reverb plug-in is available on web pages httD://www.dsoundl .coin as a part of Stomp'n FX V01.2 plug-in pack. You need a DirectX or VST compatible audio application to test this plug-in. For further information, please, contact me via email mailto:bali k@,feec.vutbr.cz.

Fig. 8. Block diagram of Model(2)

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REFERENCES

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DL (Delay Line) - in this case N should be equal to power of 2 (16,32,64,128 t4;pically). This values are required because of number of inputs to FDN (number of inputs must be power of 2). LSM (NxZ)(Linear Scaling Matrix) - 2 outputs from LSM are defined as 2 weighted sums of N inputs. The output represents Part1 on Fig. 5. MAD (Matrix and Delays) - this block contains N FIR filters specifying absorption for each DL output. These filters can specify absorption much more accurate than IIR filters used in Model(1). LSM (Nx6)(Linear Scaling Matrix) - 6 outputs from LSM are defined as 6 weighted sums of N inputs. Output from this block represents Part2 on Fig. 5. FDN (Feedback Delay Network) - N parallel comb filters with frequency dependent feedback matrix (NxN). Produces N uncorellated outputs. This is most demanding part of the algorithm, computational complexiq is much higher than computational complexity of PCF in Model(1). 6xLSM (Mxl)(Linear Scaling Matrix) - this block should producc 6 uncorelated outputs. It contains 6 LSM with M inputs and 1 output, where M is defined as nearest lower integer of N/6. MID(Mix with Delays) - this block only mix time shifted surround sounds.
The output signals C, L, R, SI, S2, S3, S4, S5, S6 are not directly prepared for reproduction. These signals must be at first properly time shifted and scaled in amplitude. Then these signals go to panning module, where is specified the direction of sound beam (see [ 111). And then

[3] Gardner, W.G. 1998. c' $er 3. Reverberation . Algorithms, in Kahrs, ivl. and Brandenburg, K. Editors. Applications of Digital Signal Processing to Audio and Acoustics. Kluwer Academic Publishers. [4] Online documentation on http://www.mlssa.corn/ [ 5 ] Schroeder. M. R., Logan, B. F. 1961. Colorless Artificial Reverberation. J. Audio Engineering Society. Vol. 9, No. 3. [6] Schroeder. M. R. 1962. Natural Sounding Artificial Reverberation. J. Audio Engineering Society. Vol. 10, No. 3. [7] Dattorro, J. 1997. Effect Design. Part 1: Reverberator and Other Filters. Journal of Audio Engineering Society. Vol. 45, No. 9. Pp. 660-684. [SI Gardner, W.G. 1992. The virtual Acoustic Room. Master Science Thesis at the MIT. [9] Jot, J.M. 1992. Etude et realisation d'un spatialisateur de sons par moddes physiques et perceptifs. Ph.D. thesis, Telecom, Paris. [lo] Rochesso, D., and Smith, J. 0. 1997. "Circulant and elliptic feedback delay networks for artificial reverberation". IEEE trans. Speech & Audio 5(1). [ 111 V. Pulkki. Virtual sound source positioning using, vector base amplitude panning. J. Audio Eng. Soc., 45(6):456--466, Jun. 1997. [ 121Dolby Digital Professional Encoding Guidelines, Dolby Laboratories Inc. 2000.

[ 11 5.1-Channel Production Guidelines, Dolby Inc. 2000. [2] Surround Sound Technologies, Dolby Inc. 2000.

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